Hi,
Today I installed version 2.9.0 and created 1 extension with my outbound routes and trunks. Unfortunately outbound call does not work.
My phone receives a “DECLINED” message from Asterisk and I don’t understand why.
Could anyone guide me about the failure please ?
I get the following output on Asterisk console:
router*CLI>
<--- SIP read from UDP:192.168.254.5:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-89e3a4df
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Home <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 304
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 56213 56213 IN IP4 192.168.254.5
s=-
c=IN IP4 192.168.254.5
t=0 0
m=audio 16416 RTP/AVP 18 0 8 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 192.168.254.5:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '995' for '995' from 192.168.254.5:5060
<--- Reliably Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-89e3a4df;received=192.168.254.5
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>;tag=as205d6115
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.9.0(1.8.4.4)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20387cf5"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.254.5:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-89e3a4df
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>;tag=as205d6115
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: Home <sip:[email protected]:5060>
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.254.5:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-61b9ae2f
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="20387cf5",uri="sip:[email protected]",algorithm=MD5,response="a57f766d866d5072a2b63e994a6dbf35"
Contact: Home <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 304
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 56213 56213 IN IP4 192.168.254.5
s=-
c=IN IP4 192.168.254.5
t=0 0
m=audio 16416 RTP/AVP 18 0 8 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.254.5:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '995' for '995' from 192.168.254.5:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.254.5:16416
Looking for 3616246 in from-internal (domain 192.168.254.254)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-61b9ae2f;received=192.168.254.5
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.4.4)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [3616246@from-internal:1] Macro("SIP/995-0000000e", "user-callerid,LIMIT,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/995-0000000e", "AMPUSER=995") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/995-0000000e", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/995-0000000e", "1?Set(REALCALLERIDNUM=995)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/995-0000000e", "AMPUSER=995") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/995-0000000e", "AMPUSERCIDNAME=Ev Giris") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/995-0000000e", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/995-0000000e", "AMPUSERCID=995") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/995-0000000e", "CALLERID(all)="Ev Giris" <995>") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/995-0000000e", "0?limit") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/995-0000000e", "1?Set(GROUP(concurrency_limit)=995)") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/995-0000000e", "1?continue") in new stack
-- Goto (macro-user-callerid,s,24)
-- Executing [s@macro-user-callerid:24] Set("SIP/995-0000000e", "CALLERID(number)=995") in new stack
-- Executing [s@macro-user-callerid:25] Set("SIP/995-0000000e", "CALLERID(name)=Ev Giris") in new stack
-- Executing [s@macro-user-callerid:26] Set("SIP/995-0000000e", "CHANNEL(language)=en") in new stack
-- Executing [3616246@from-internal:2] Set("SIP/995-0000000e", "MOHCLASS=default") in new stack
-- Executing [3616246@from-internal:3] Set("SIP/995-0000000e", "_NODEST=") in new stack
-- Executing [3616246@from-internal:4] Macro("SIP/995-0000000e", "record-enable,995,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/995-0000000e", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/995-0000000e", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/995-0000000e", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing [s@macro-record-enable:14] GotoIf("SIP/995-0000000e", "0?IN") in new stack
-- Executing [s@macro-record-enable:15] ExecIf("SIP/995-0000000e", "1?MacroExit()") in new stack
-- Executing [3616246@from-internal:5] Macro("SIP/995-0000000e", "dialout-trunk,2,3616246,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/995-0000000e", "DIAL_TRUNK=2") in new stack
== Spawn extension (macro-dialout-trunk, s, 2) exited non-zero on 'SIP/995-0000000e' in macro 'dialout-trunk'
== Spawn extension (from-internal, 3616246, 5) exited non-zero on 'SIP/995-0000000e'
-- Executing [h@from-internal:1] Hangup("SIP/995-0000000e", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/995-0000000e'
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-61b9ae2f;received=192.168.254.5
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>;tag=as2aaa4f08
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.4.4)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.254.5:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-61b9ae2f
From: Home <sip:[email protected]>;tag=18185e19f7db1ecfo0
To: <sip:[email protected]>;tag=as2aaa4f08
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="20387cf5",uri="sip:[email protected]",algorithm=MD5,response="a57f766d866d5092a2b63e994a6dbf35"
Contact: Home <sip:[email protected]:5060>
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0